November 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Capture SIP methods and their contents in a log file
by Sam Montour 05 Nov '13

05 Nov '13

Hi, I am implementing a couple of web applications, one to gather accounting and CDR data for billing information about a call. The second web application is to display a SIP call flow with the SIP methods and their contents. I used XML CDR log files to gather everything I needed for the billing application. But for the SIP methods and their contents, I am not sure where to retrieve such info. I know from previous experience, I used the FS command line to turn the SIP trace on using "sofia profile <profile_name> siptrace on". This however displays the output to fs_cli command window. Is there a way I can retrieve SIP methods and contents from a log file like we can with xml cdr log files? You feedback is very much appreciated. Thanks. Sami

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Re: [Freeswitch-users] Scalability and Bridge
by DanB 05 Nov '13

05 Nov '13

Hey Kristian, I was talking exactly about that case: an app doing 2 sessions and bridging them together with reinvite. DanB On 05.11.2013 18:23, freeswitch-users-request(a)lists.freeswitch.org wrote: > Reinvite only updates an existing session. You cannot reinvite to a > new/different endpoint.

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Jira not accessible
by Mario G 05 Nov '13

05 Nov '13

In case no one reported it Jira will not display first page and times out this morning. Mario G

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Bypass media scenario
by Grant Bagdasarian 05 Nov '13

05 Nov '13

Hello, Consider the following flow: UAC --- FreeSWITCH (SBC) --- Kamailio/MediaProxy ---- UAS FS only bridges the incoming leg with an outgoing one towards Kamailio. Is it possible to have the RTP stream flow between the UAC and UAS through the MediaProxy, even when bypass media is disabled on FS? If not, is it possible to have Kamailio control this in some way? Note, that I do want the signaling to still pass through FS at all times.

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Perl equivalent dialplan
by mehroz 05 Nov '13

05 Nov '13

I can get the registration status of a user in XML dialplan with : <condition field="${sofia_contact(internal/${destination_number}(a)192.168.1.5)}" expression="error/user_not_registered"> ... and route the call accordingly. I am writing a perl script and calling it in XML with <action application="perl" data="/usr/local/src/test.pl/> and want to get the same registration status from the same CHANNEL VARIABLE. I can get CHANNEL VARIABLE using $session->getVariable("variable"); but cannot get ${sofia_contact(internal/${destination_number}(a)192.168.1.5)} value with this procedure. I believe this is not actually a CHANNEL VARIABLE , but I cant find ant other way of retrieving this information within my script. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Perl-equivalent-dialplan-tp75… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Re: [Freeswitch-users] Scalability and Bridge
by DanB 05 Nov '13

05 Nov '13

REFER is another good idea. The only issue is that you will need to trust the originator handling it properly vs establishing the legs yourself and reinvite which could be more on your control. Cheers, DanB On 05.11.2013 16:02, freeswitch-users-request(a)lists.freeswitch.org wrote: > A REFER (transfer) would be more appropriate in this scenario.

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Re: [Freeswitch-users] Raspberry Pi, mod_gsmopen GSM dongle
by Sergey Zhuravlov 05 Nov '13

05 Nov '13

Hi! Not so bad FS I still work with the module gsm_open, sometimes it is not clear why the device is heavily loaded. load average: 4.07, 3.99, 3.96 Now I want to watch CPU module with or without gsm_open. I want to collect data using sar and show to the public. The sound quality is not very much, but like a normal. use Raspbian Linux raspberrypi 3.6.11+ #538 PREEMPT Fri Aug 30 20:42:08 BST 2013 armv6l GNU/Linux For I am not afraid that the long compile, just to work with dignity;-) From: Johannes Stads <stadsj(a)yahoo.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Sun, 3 Nov 2013 11:39:49 -0800 (PST) > Subject: Re: [Freeswitch-users] Raspberry Pi, mod_gsmopen GSM dongle > Sergey I tend to agree with the statement in your first post, it's a shame > that Asterisk works better > in your setup. But i think there the problem is not in Freeswitch. The > problem is > in gsmopen module, in its features and support. > I made a question about gsmopen back in september, and did not receive any > answer. Gsmopen is in > a beta state, we should not require any stable operation of FS with > gsmopen module in a production > environment. > I have a long experience with Asterisk, and decided to give a try to > Freeswitch to see its superior > performance. I started using a linux virtual machine. Then tried it also > with BeagleBone Black which is > just a bit more powerful beast than PI. All compiled well, including the > latest version of FS. > Building of a mod_gsmopen was a bit of a pain, but finally worked too. > What linux flavor were you using with FS on PI? > In my setup I used bare minimum config, and again can confirm that FS is a > superior platform with > plenty of advanced features, though its concept is more difficult for > understanding initially. > Although I was able to send SMS and make calls out of the box, gsmopen > worked highly unstable and > sound quality was not good, to say the least. > > To prove my concept, today i took a fresh SD card to put Debian Wheezy on > it and installed Asterisk > with chan_dongle. The platform used was BBB. > What can i say, the experience was different. To compare, FS build time on > BBB/Debian/SD card was > such: > ./configure (about 1h) > make (about 2h) > Asterisk was up and running in about 30 min. > chan_dongle was built smoothly without a need to tweak libraries. > > In my FS setup gsmopen did not show a claimed 'auto discovery' feature and > some exercise with > UDEV rules was required to make sure that modems are loaded in right order. > In chan_dongle it is nicely done out of the box. > > Overall, chan_dongle is much more advanced piece of software, I want to > use this opportunity to thank its > author. If your goal is calling to mobiles, you have to consider > chan_dongle against gsmopen, > and not FS against *. > > Cheers, > Jo > > ------------------------------ > -------------- > On Wed, 10/30/13, Sergey Zhuravlov <4orbit(a)gmail.com> wrote: > > Subject: Re: [Freeswitch-users] Raspberry Pi, mod_gsmopen GSM dongle > To: freeswitch-users(a)lists.freeswitch.org > Date: Wednesday, October 30, 2013, 4:16 AM > > I am glad to > tell you that: > 1) mod_gsmopen and e1550 work on Raspberry Pi (incoming, > outgoing SMS in both directions). > > 2) and the stable version FS and the latest FS compile on Raspberry Pi, > but it's a long time. > > 3) The latest version does not run on my raspberries. > > Need advice on optimizing the FS for raspberry Pi, as low-productivity > device. > > > Can any startup options? > Maybe someone knows what modules you can disable resource-intensive. > > Something else? > > > Does somebody > have a positive experience with > Raspberry > Pi, mod_gsmopen GSM > dongle ??? > > I use Raspberry Pi > B and e1550. > > It's a shame that > with an asterisk all > works fine -- RasPBX. http://www.raspberry-asterisk.org/ > > > > But I prefer the > FreeSWITCH ;-) > On the > basis of experience, I thought > that PS will work best > for this modest > hardware faster and > require fewer resources. > > > > But it turns out it is not. > Asterisk with a web > server, MySQL database, and > other, less load > than FS. > > > In this connection, other > parameters may be used > starting FS to save > resources? > > > And the main > question! Silence an > incoming call on 5000 > (IVR) as well as an > outgoing in gsm > network. Those are not > sound. > > > > -- > WBR, Sergey > > GTALK/JABBER:4orbit@gmail.com > > > > -- > WBR, Sergey > > GTALK/JABBER:4orbit@gmail.com > > > -----Inline Attachment Follows----- > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- WBR, Sergey GTALK/JABBER:4orbit@gmail.com -- WBR, Sergey GTALK/JABBER:4orbit@gmail.com

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Re: [Freeswitch-users] Scalability and Bridge
by DanB 05 Nov '13

05 Nov '13

Valter, If the call is already established and you want to move it into a conference knowing the box where the conference is I guess you need own application logic which will call the bridge from the box where your call is active and do a reinvite then putting the party you want to bridge with the bridge itself. DanB

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Re: [Freeswitch-users] Scalability and Bridge
by Valter Cardoso 05 Nov '13

05 Nov '13

I think 302 Redirect only works at call establishment time. My problem is the call is already active in a FS and I need to bridge it with a call in another FS (or add it to a conference in another FS). For this, I think 302 will not help. Thanks, Valter

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481 Call Does Not Exist
by Miha 04 Nov '13

04 Nov '13

Hi, I need a little help with figuring out why FS sends "481". This is happing when call has been forward (302). All calls are relayed throught opensips as I am using it for registrations and load_balancing. http://pastebin.freeswitch.org/21555 tnx! miha

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